I have an application on top of WebRTC. If a peer (A) clicks on the link to the application and starts it in the middle of a phone call, then that peer (A) is getting connected to peer (B) successfully. Even after ending the call, Peer A is not able to hear Peer B whereas Peer-B is able to see the video as well as hear the audio of Peer - A. The application works perfectly if the PeerConnection was created without any ongoing phone call.
I have tried to restart/createOffer the ICE and tried to acquire the user-media again and replace the tracks. But nothing works.
- Is there a way I can find out in browser that there is an ongoing call so that I can prevent the customer from even initiating the RTCPeerConnection?
- How to recover from this situation of no-audio at PeerA even after hanging up the call